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gerrit-public.fairphone.software
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platform
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frameworks
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base
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20339b24394be06c5bfe5157a4b35adc48f665eb
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voip
b7a76e8
am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 13 years ago
d7a724e
Issue 4157048: mic gain for VoIP/SIP calls.
by Eric Laurent
· 13 years ago
b5c72ea
Making it possible to call SIP calls with special allowed chars.
by Magnus Strandberg
· 13 years ago
2ba92c7
do not merge bug 3370834 Cherrypick from master
by Jean-Michel Trivi
· 14 years ago
14b6d06
Merge changes Ib70e0cf2,I0691cd70 into gingerbread
by Hung-ying Tyan
· 14 years ago
f268a2f
Add auth. username in SipProfile.
by Chung-yih Wang
· 14 years ago
f0bb1ce
SipService: registers broadcast receivers on demand.
by Hung-ying Tyan
· 14 years ago
0c01e6e
SipService: release wake lock for cancelled tasks.
by Hung-ying Tyan
· 14 years ago
3cf7137
RTP: Send silence packets on idle streams for every second.
by Chia-chi Yeh
· 14 years ago
5bd3782
Check if VoIP API is supported in SipManager.
by Hung-ying Tyan
· 14 years ago
a936b25
Remove SIP realm/domain check
by Hung-ying Tyan
· 14 years ago
58ee2ac
Check port in create peer's SIP profile.
by Hung-ying Tyan
· 14 years ago
f053292
Fix SIP bug of different transport/port used for requests.
by Chung-yih Wang
· 14 years ago
e2abd10
Merge "Set AudioGroup mode according to audio settings" into gingerbread
by Hung-ying Tyan
· 14 years ago
06e8cdc
Fix race between ending and answering a SIP call.
by Hung-ying Tyan
· 14 years ago
fa81463
Set AudioGroup mode according to audio settings
by Hung-ying Tyan
· 14 years ago
4189d99
Do not suppress error feedback during a SIP call.
by Hung-ying Tyan
· 14 years ago
349f350
Merge "Correct SipService.isOpened() implementation." into gingerbread
by Hung-ying Tyan
· 14 years ago
d9e1230
Merge "Notify SipSessions before closing SIP stack." into gingerbread
by Hung-ying Tyan
· 14 years ago
0bba953
Merge "Throw proper exceptions in SipManager" into gingerbread
by Hung-ying Tyan
· 14 years ago
bd399b0
Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread
by Chia-chi Yeh
· 14 years ago
8d1b2a1
Throw proper exceptions in SipManager
by Hung-ying Tyan
· 14 years ago
262cdfc
Correct SipService.isOpened() implementation.
by Hung-ying Tyan
· 14 years ago
02b1d68
docs: revise javadocs for sip
by Scott Main
· 14 years ago
0c7d306
RTP: Pause echo suppressor when far-end volume is low.
by Chia-chi Yeh
· 14 years ago
5d0c5cf
Notify SipSessions before closing SIP stack.
by Hung-ying Tyan
· 14 years ago
60c45d0
Clean up pending sessions on incoming call in SipService
by Hung-ying Tyan
· 14 years ago
703aae0
Merge "RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples." into gingerbread
by Chia-chi Yeh
· 14 years ago
8a68b52
RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples.
by Chia-chi Yeh
· 14 years ago
e87b644
Add permission requirements to SipAudioCall and SipManager javadoc.
by Hung-ying Tyan
· 14 years ago
9b449e5
Remove ringtone API from SipAudioCall.
by Hung-ying Tyan
· 14 years ago
4944fdd
Periodically scan wifi when wifi is not connected and wifi lock is
by Chung-yih Wang
· 14 years ago
0a6e717
Handle dialing a SIP call to self.
by Hung-ying Tyan
· 14 years ago
431bb22
Reduce logging.
by Joe Onorato
· 14 years ago
b4116c0
Fix the incorrect environment variable name for the thread pool size.
by Chung-yih Wang
· 14 years ago
45bd830
Merge "Uncomment SIP/VOIP feature check in SipManager." into gingerbread
by John Huang
· 14 years ago
a0cdfbf
Uncomment SIP/VOIP feature check in SipManager.
by Hung-ying Tyan
· 14 years ago
66cc535
Set the thread pool size of NIST sip stack to one.
by Chung-yih Wang
· 14 years ago
28f63c0
SipService: add wake lock for incoming INVITE packets.
by Hung-ying Tyan
· 14 years ago
bd57eea
SipService: add wake lock for multiple components.
by Hung-ying Tyan
· 14 years ago
4f8fd10
Make SipService listen to WIFI state change events.
by Hung-ying Tyan
· 14 years ago
f1b1eec
Merge "SipService: mScreenOn is flipped to wrong value." into gingerbread
by Hung-ying Tyan
· 14 years ago
d6fc979
SipService: mScreenOn is flipped to wrong value.
by Hung-ying Tyan
· 14 years ago
ebc886c
Fix SipSessionGroup from throwing ConcurrentModificationException
by Hung-ying Tyan
· 14 years ago
685b61b
SipService: fix a missing switch-case break.
by Hung-ying Tyan
· 14 years ago
692cac9
SipHelper: add debug log for challenge responses.
by Hung-ying Tyan
· 14 years ago
c7fda18
Do not release the wifi lock if the screen is off.
by Chung-yih Wang
· 14 years ago
aa562ff
SipService: add permission check for using API
by Hung-ying Tyan
· 14 years ago
08faac3
Unhide SIP API.
by Hung-ying Tyan
· 14 years ago
4483232
Suppress harder for echo without affecting the volume of real voice.
by Chung-yih Wang
· 14 years ago
9db99a4
Make SipService broadcast SIP_SERVICE_UP when it's up.
by Hung-ying Tyan
· 14 years ago
16c29bd
Merge "SIP: Fix busy authentication loop." into gingerbread
by Hung-ying Tyan
· 14 years ago
ee8a884
SIP: Fix busy authentication loop.
by Hung-ying Tyan
· 14 years ago
fb116fb
Misc fixes for sim-eng build.
by Chung-yih Wang
· 14 years ago
718e003
Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread
by Hung-ying Tyan
· 14 years ago
c6548fd9
SIP: add SERVER_UNREACHABLE error code.
by Hung-ying Tyan
· 14 years ago
323d367
SipService: supply PendingIntent when open a profile.
by Hung-ying Tyan
· 14 years ago
4fc04f1
Merge "RTP: Add a baseline echo suppressor." into gingerbread
by Chia-chi Yeh
· 14 years ago
a8a1009
RTP: Add a baseline echo suppressor.
by Chia-chi Yeh
· 14 years ago
9ea96c6
SIP: minor fixes.
by Hung-ying Tyan
· 14 years ago
274e3b5
Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread
by Chia-chi Yeh
· 14 years ago
063d02b
Merge "SipService: turn off verbose logging" into gingerbread
by Hung-ying Tyan
· 14 years ago
67ecb5b
RTP: Start AudioRecord before AudioTrack to avoid being disabled.
by Chia-chi Yeh
· 14 years ago
b031957
SipService: turn off verbose logging
by Hung-ying Tyan
· 14 years ago
21ae1ad
RTP: Minor fixes with polishing.
by Chia-chi Yeh
· 14 years ago
d29e075
Merge "Add uri field to SipManager.ListenerRelay" into gingerbread
by Hung-ying Tyan
· 14 years ago
9e1d308
Add uri field to SipManager.ListenerRelay
by Hung-ying Tyan
· 14 years ago
dfd1484
Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread
by Chia-chi Yeh
· 14 years ago
3520bd4
RTP: Adjust the jitter buffer to 512ms.
by Chia-chi Yeh
· 14 years ago
6a53489
SipService: add UID check.
by Hung-ying Tyan
· 14 years ago
0a537b7
Merge "RTP: Enable AMR codec." into gingerbread
by Chia-chi Yeh
· 14 years ago
2365b78
Merge "SIP: misc fixes." into gingerbread
by Hung-ying Tyan
· 14 years ago
f88fc1f
RTP: Enable AMR codec.
by Chia-chi Yeh
· 14 years ago
fb3a98b
SIP: misc fixes.
by Hung-ying Tyan
· 14 years ago
f4ae942
RTP: Enable GSM-EFR codec.
by Chia-chi Yeh
· 14 years ago
fe52989
RTP: Revise the workaround of private addresses and fix bugs.
by Chia-chi Yeh
· 14 years ago
e006e4d
Merge changes Iae1913fb,I38dbefef into gingerbread
by Chia-chi Yeh
· 14 years ago
a6f950c
RTP: Enable GSM codec.
by Chia-chi Yeh
· 14 years ago
78c11b3
RTP: Refactor out G711 codecs into another file.
by Chia-chi Yeh
· 14 years ago
320cdcb
Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread
by Chia-chi Yeh
· 14 years ago
9083c84
RTP: Delay the initialization of AudioTrack and AudioRecord.
by Chia-chi Yeh
· 14 years ago
6057cd0
SIP: Feedback any provisional responses in addition to RING
by Hung-ying Tyan
· 14 years ago
624d5b4
SIP: add DisconnectCause.SERVER_ERROR
by Hung-ying Tyan
· 14 years ago
7e54ef7
Move SipService out of SystemServer to phone process.
by Hung-ying Tyan
· 14 years ago
fd144d7
Merge "SipAudioCall: remove SipManager dependency." into gingerbread
by Hung-ying Tyan
· 14 years ago
00a2206
SipService: handle cross-domain authentication error
by Hung-ying Tyan
· 14 years ago
bd22942
Fix the unhold issue especially if one is behind NAT.
by Chung-yih Wang
· 14 years ago
3a4197e
SipAudioCall: remove SipManager dependency.
by Hung-ying Tyan
· 14 years ago
658bec9
SDP: remove dead code.
by Chia-chi Yeh
· 14 years ago
84a357b
Refactoring SIP classes to get ready for API review.
by Hung-ying Tyan
· 14 years ago
0b7d6de
Fix the build.
by repo sync
· 14 years ago
84f7f6b
SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.
by repo sync
· 14 years ago
e6c0c10
SDP: Add a simple class to help manipulate session descriptions.
by Chia-chi Yeh
· 14 years ago
7a69aef
RTP: Add log throttle for "no data".
by repo sync
· 14 years ago
4033a67
RTP: Update native part to reflect the API change.
by Chia-chi Yeh
· 14 years ago
37adc52
RTP: Add two getters to retrieve the current configuration from AudioStream.
by Chia-chi Yeh
· 14 years ago
32e106b
RTP: Extend codec capability and update the APIs.
by Chia-chi Yeh
· 14 years ago
8544560
SipPhone: fix missing-call DisconnectCause feedback
by Hung-ying Tyan
· 14 years ago
9796379
SIP: convert enum to static final int.
by Hung-ying Tyan
· 14 years ago
c4b8747
SIP: add config flag for wifi-only configuration.
by Hung-ying Tyan
· 14 years ago
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